external/webrtc.git -

Age	Commit message (Expand)	Author
2015-10-07	Use suffixed {uint,int}{8,16,32,64}_t types.	Peter Boström
2015-08-24	Update a ton of audio code to use size_t more correctly and in general reduce	Peter Kasting
2015-08-13	In PeerConnectionTestWrapper, put audio input on a separate thread.	deadbeef
2015-06-11	Reformat existing code. There should be no functional effects.	Peter Kasting
2015-06-11	Match existing type usage better.	Peter Kasting
2015-02-09	Remove USE_WEBRTC_DEV_BRANCH.	pbos@webrtc.org
2015-01-29	Remove ChangeUniqueID.	tommi@webrtc.org
2015-01-20	Update libjingle license statements at top of talk files for consistency	jlmiller@webrtc.org
2014-12-15	Use int64_t for milliseconds more often, primarily for TimeUntilNextProcess.	pkasting@chromium.org
2014-09-11	Mark all virtual overrides in the hierarchy of Module as virtual and OVERRIDE.	henrik.lundin@webrtc.org
2014-07-29	(Auto)update libjingle 72097588-> 72159069	buildbot@webrtc.org
2014-07-14	(Auto)update libjingle 71107853-> 71115715	buildbot@webrtc.org
2014-06-05	Fix the chain that propagates the audio frame's rtp and ntp timestamp including:	wu@webrtc.org
2014-05-19	Add interface to propagate audio capture timestamp to the renderer.	wu@webrtc.org
2013-10-22	Add CriticalSection to fakeaudiocapturemodule to protect the variables which ...	wu@webrtc.org
2013-07-10	Adds trunk/talk folder of revision 359 from libjingles google code to	henrike@webrtc.org